11 Sep
2003
11 Sep
'03
7:08 a.m.
I have tried to work through your suggestions. Please see my responses.
I appreciate your attentivness and help.
Stephen
>1) Having issues with correctly increasing the
number of children. If
I
What problems do you have with high number of
connections in my.cnf?
Most of the documentation that I had seen had my use max-connections =
x. I finally found a suggestion to use set-variable=max-connections=x
and it worked.
>2) I only want the call to fail to route 2 for
calls that terminate in
>my network. I don't want calls leaving the network to try to go to
>voicemail after the time hits.
Sure. What is the question, script? If so, that's
easy: set
t_on_failure
only from within the uri==myself condition.
Unless I am mistaken, that will only help calls originating outside of
my network. If I place a call to 18885551212, even if it's not a local
destination, the uri still looks like sip:18885551212@209.242.10.153 and
matches uri==myself. I did place this portion inside the myself check
and it still tries to transfer to vm after the time expires.
Actually, most of your script should be within this
condition. If
a request for other domain comes (i.e., uri==msyelf does not hold),
you just forward using t_relay and that's it. All the script
processing,
uri rewriting, redirection makes only sense if you
"own" the request.
Thanks, I fixed that.
>3) I can't seem to figure out how to format the
rewrite/append/etc to
>trigger the voicemail call. Maybe I don't understand the routing
loop.
>Can someone take a look at the config and give me
some ideas?
The "user-offline" part is I guess ok. There
are some nits though:
- you ignore non-VoIP messages to off-line users -- is an instant
MESSAGE comes, you just break. you should indicate the status
sip-wise instead. Use sl_send_reply("404","not found");
- I don't know what the append_branch in your script is good for,
but I don't think you need it.
This is the part that I really need help with! When the call timer
fails, the call goes to the route[1]. How do I get it into voice mail
from that point?
>5) I'm not seeing the activity/routing logs
going anywhere. If I run
>ser manually, I see information on the stderr, but nothing in the
>database or logs in terms of the connections.
If you start manually, force messages and see them on
your console,
then everything is all right, isn't it? If you start from an init
script and keep forcing use of stderr, you will obviously will not
see anything -- you need to turn stderr off and watch syslog output.
This did send the syslog to the messages file, but where are the call
transaction records?
Here is my lastest config.
modparam("tm", "fr_inv_timer", 5 )
modparam("tm", "fr_timer", 10 )
modparam("usrloc", "db_mode", 2)
#modparam("auth_db",
"db_url","sql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("voicemail",
"db_url","sql://ser:heslo@localhost/ser")
modparam("rr", "enable_full_lr", 1)
# -- acc params --
modparam("acc", "report_ack", 1)
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1 )
modparam("acc", "db_flag", 1 )
modparam("acc", "log_missed_flag", 3 )
modparam("acc", "db_missed_flag", 3 )
# report to syslog: From, i-uri, status, digest id
modparam("acc", "log_fmt", "fisu" )
# ------------------------- request routing logic -------------------
alias="test.net"
alias="209.242.10.153"
route{
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Wow -- Message too large");
break;
};
if (loose_route()) { t_relay(); break; };
if (method=="INVITE") {record_route();};
# account completed transactions via syslog
setflag(1);
if (uri==myself) {
if (src_ip==66.155.138.5) {
log("gateway-originated request");
} else {
if (method=="REGISTER") {
log(1,"authenticating");
if (!www_authorize("test.net",
"subscriber")) {
www_challenge("test.net", "0");
break;
};
save("location");
break;
};
};
if (uri=~"sip:voicemail\+@") {
log(1,"call matches voicemail");
t_newtran();
t_reply("100", "trying -- just a second");
if (!vm("/tmp/am_fifo","announcement")) {
t_reply("500", "SEMS error");
};
break;
};
if (uri=~"sip:2[0-9]+@.*") {
log(1,"call matches local number");
if (!lookup("location")) {
log(1,"failed lookup");
if (method=="INVITE" || method=="ACK") {
t_newtran();
t_reply("100", "trying -- one
moment");
if(!vm("/tmp/am_fifo","voicemail")){
t_reply("500", "SEMS
error");
};
break;
};
};
} else {
rewritehostport("66.155.138.5:5060");
};
t_on_failure("1");
if (!t_relay()) {
sl_reply_error();
break;
};
};
}
failure_route[1]{
log(1,"call sent to voicemail due to no answer\n");
if (uri=~"sip:2[0-9]+@.*") {
if (method=="INVITE" || method=="ACK") {
#WHAT DO I PUT HERE TO TRIGGER THE VM?
t_relay();
};
break;
};
}